The necessity of digital communication over existing communications networks, such as telephone companies, has necessitated the use of increasingly complex and efficient modulators/demodulators (modems). Because most communication systems were originally designed as analog systems, conventional modems have always been limited by the necessity of analog capabilities. Consequently, such modems operate as if the entire communications network, such as a public switch telephone network (PSTN), is an analog system, even though much of the communication throughout the PSTN is digital in nature.
FIG. 3 is a block diagram of a communications system, which includes a digital public switch telephone network (PSTN) 40. Remote or client modem 10 communicates with a server modem installation 20 through digital PSTN 40. In order to communicate with the digital portion 40 of the PSTN, it may be necessary to communicate over an analog link 50. Accordingly, digital information at client modem 10 is converted to analog signals, transmitted over analog link 50 (generally designated as a local loop), and transmitted to PSTN circuitry 33, which is a hybrid that converts the 2 wire bi-directional local loop signal to 4 wire transmit and receive pair of signals and 30, which is a network interface that does the conversion from the analog loop signals to digital trunk signals using ADC 31, and also does the PCM encoding in case a mu-law or A-law CODEC interface is used and transmitted into digital portion 40 of PSTN 40.
Such systems are characterized by the use of an analog-to-digital converter (ADC) 31, which is used to digitize analog signals for transmission to a digital portion of the PSTN 40. Two types of quantization systems may be used in such systems, mu-law and A-law. Both systems encompass signal compression algorithms to maximize the dynamic range of an analog signal that could be represented by 8 bits. Both of these systems are used to optimize the PSTN for traditional voice communications.
Unfortunately, the wide dynamic range of normal speech does not lend itself well to efficient linear digital encoding. Digitizing is done in both systems by limiting the Signal to Quantization noise of the signals thus reducing the bandwidth required for each call. These encoding systems impose significant limitations on data communications.
Client modem 10 may typically transmit data using a digital-to-analog converter (DAC) 15 to convert digital signals into analog signals. Eventually, the analog signal is received and converted back to digital form by an analog-to-digital converter (ADC) 31 somewhere in the PSTN 40. When the analog signal levels transmitted by DAC 15 in client modem 10 do not accurately correspond to the quantization intervals used by receiving ADC 31, the data transmitted may not be properly converted back into the exact digital form originally sent. If receiving ADC 31 incorrectly converts an analog signal transmitted, receiving modem (e.g., server modem 20) may not receive the same data which was transmitted. The same is true in the downstream direction of server digital modem 20 to client analog modem 10.
In order to avoid such communications errors, certain error-checking protocols may be used. However, such protocols may require retransmission of corrected data, thereby reducing the rate at which data can be transmitted. The result is greatly reduced digital efficiency for the PSTN constituting the digital communications system.
Under the V.90 standard, a modem at a client computer may request that a server modem transmit a learning sequence in order to characterize the data channel. The V.90 specification does not specify a particular learning sequence and thus each manufacturer or communications system operator may use an unique learning sequence to characterize the data channel.
In some T1 lines, a telephone company (Telco) may use one of a number of time-division multiplexed channels for transmission of control data. However, in other instances, a technique of "robbed bits" (RBS) may be used to transmit control data for use by the Telco. In the robbed bit signaling technique, a least significant bit of one or more slots is used during initial connection to transmit control data from one portion of the Telco to another indicating numbered dialed, hang up, and the like. During data transmission, this same least significant bit may be set to a predetermined value (e.g., 0 or 1), or may be toggled.
In traditional voice communications signalling, Robbed Bit techniques produced few problems, as arbitrarily changing the least significant bit of a digitized audio signal does not produce any noticeable audio artifacts in the resultant analog audio signal. However, in a digital signalling environment, such legacy techniques as Robbed Bit Signalling create problems for digital data transmission. Arbitrarily changing the least significant bit in a data stream may produce a digital data transmission error.
The wide variety of digital system characteristics means digital impairments such as RBS and pad variations, as well as CODEC types which would modify the digital information the server outputs before it is converted to analog information that the client receives. Apart from these there could be analog impairments such as line distortion, noise and Inter-Modulation Distortion (IMD) which could create conditions for greater communications errors. Consequently, additional techniques are necessary to properly measure the pad and accommodate any RBS such that received signals can be properly adjusted to reflect the reality of those signals as sent, rather than theoretical values, which may prove to be incorrect.